hello Matthieu,
no my speaker is not a dipole aned it is not close to lateral wall. Could
it be a reflexion on the floor that you observed?
It is true that there is no carpet on the path from the left speaker to my
listening position.
2015-05-08 9:04 GMT+02:00 Matthieu VM <matvm@...>:
> Hello Vincent,
>
> On the left channel, the second peak of the impulse (with 1ms delay) is as
> strong as the first one.
> Actually when I tried to open the impulse in REW, it aligned on the second
> peak...
> This results in irregularities on the response of windowed signal, like
> the Unsmoothed1ms plot.
>
> Is your speaker a dipole, or very close to lateral wall ?
>
> Matthieu
>
>
>
> Le Lundi 4 mai 2015 21h11, Vincent Lejay <vincent.lejay@...> a
> écrit :
>
>
> hi all
>
> Back Home!
>
> here are 2 links to my dropbox (I hope you can have access to it!)
>
> first the Impulse response of my system (computed by Align2) + my DRC
> config file(s):
> https://www.dropbox.com/sh/wvjp38mzahoe77s/AAAFhDU7FvkctmVTapxhXVxCa?dl=0
>
> then the octave graphs based on these IR
> https://www.dropbox.com/sh/oi67m08g0wmmkhy/AAAumUlbqkvhG50F2cGA_K6Sa?dl=0
>
> Do you see anything wrong with all this?
>
> Regards
> Vincent
>
> 2015-04-28 9:19 GMT+02:00 Denis Sbragion <d.sbragion@...>:
>
> Hello Vincent,
>
> On Mon, April 27, 2015 20:25, Vincent Lejay wrote:
> ...
> > When I say that the measure of the corrected signal is different from the
> > theoretical one, I mainly speak of the "time correction". The frequency
> > response seems good, but thevarious ocatve graphs showing the IR, step
> > reponse... seem more dirty than the original.
>
> it should be the same thing. As long as you don't move the mic the measured
> and simulated IR should be pretty much identical.
>
> Here it is a test I did more than 10 years ago:
>
> http://servizi.infotecna.it/MisVsSim/MisVsSim.png
>
> This has been done using MLS instead of the log sweep as the measurement
> method, and a really cheap SoundBlaster PCI 128 soundcard.
>
> It's not easy to see it but there are actually two tracks there, one blue
> and
> the other one green, almost perfectly superimposed. I don't even remember
> which is the measurements and which is the simulation. Consider that MLS is
> much less accurate than the log sweep, so by using the log sweep and some
> good
> equipment the match should be even better than that.
>
> Bye,
>
> --
> Dr. Denis Sbragion
> InfoTecna
> Tel: +39 0362 805396, Fax: +39 0362 805404
> URL: http://www.infotecna.it
>
>
>
> ------------------------------------------------------------------------------
> One dashboard for servers and applications across Physical-Virtual-Cloud
> Widest out-of-the-box monitoring support with 50+ applications
> Performance metrics, stats and reports that give you Actionable Insights
> Deep dive visibility with transaction tracing using APM Insight.
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> _______________________________________________
> Drc-fir-users mailing list
> Drc-fir-users@...
> https://lists.sourceforge.net/lists/listinfo/drc-fir-users
>
>
>
>
> ------------------------------------------------------------------------------
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> Widest out-of-the-box monitoring support with 50+ applications
> Performance metrics, stats and reports that give you Actionable Insights
> Deep dive visibility with transaction tracing using APM Insight.
> http://ad.doubleclick.net/ddm/clk/290420510;117567292;y
>
> _______________________________________________
> Drc-fir-users mailing list
> Drc-fir-users@...
> https://lists.sourceforge.net/lists/listinfo/drc-fir-users
>
>
>
>
> ------------------------------------------------------------------------------
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> Widest out-of-the-box monitoring support with 50+ applications
> Performance metrics, stats and reports that give you Actionable Insights
> Deep dive visibility with transaction tracing using APM Insight.
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> _______________________________________________
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>
>

Hello Vincent,
On the left channel, the second peak of the impulse (with 1ms delay) is as strong as the first one.Actually when I tried to open the impulse in REW, it aligned on the second peak...This results in irregularities on the response of windowed signal, like the Unsmoothed1ms plot.
Is your speaker a dipole, or very close to lateral wall ?
Matthieu
Le Lundi 4 mai 2015 21h11, Vincent Lejay <vincent.lejay@...> a écrit :
hi all
Back Home!
here are 2 links to my dropbox (I hope you can have access to it!)
first the Impulse response of my system (computed by Align2) + my DRC config file(s):
https://www.dropbox.com/sh/wvjp38mzahoe77s/AAAFhDU7FvkctmVTapxhXVxCa?dl=0
then the octave graphs based on these IR
https://www.dropbox.com/sh/oi67m08g0wmmkhy/AAAumUlbqkvhG50F2cGA_K6Sa?dl=0
Do you see anything wrong with all this?
Regards
Vincent
2015-04-28 9:19 GMT+02:00 Denis Sbragion <d.sbragion@...>:
Hello Vincent,
On Mon, April 27, 2015 20:25, Vincent Lejay wrote:
...
> When I say that the measure of the corrected signal is different from the
> theoretical one, I mainly speak of the "time correction". The frequency
> response seems good, but thevarious ocatve graphs showing the IR, step
> reponse... seem more dirty than the original.
it should be the same thing. As long as you don't move the mic the measured
and simulated IR should be pretty much identical.
Here it is a test I did more than 10 years ago:
http://servizi.infotecna.it/MisVsSim/MisVsSim.png
This has been done using MLS instead of the log sweep as the measurement
method, and a really cheap SoundBlaster PCI 128 soundcard.
It's not easy to see it but there are actually two tracks there, one blue and
the other one green, almost perfectly superimposed. I don't even remember
which is the measurements and which is the simulation. Consider that MLS is
much less accurate than the log sweep, so by using the log sweep and some good
equipment the match should be even better than that.
Bye,
--
Dr. Denis Sbragion
InfoTecna
Tel: +39 0362 805396, Fax: +39 0362 805404
URL: http://www.infotecna.it
------------------------------------------------------------------------------
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Performance metrics, stats and reports that give you Actionable Insights
Deep dive visibility with transaction tracing using APM Insight.
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hi all
Back Home!
here are 2 links to my dropbox (I hope you can have access to it!)
first the Impulse response of my system (computed by Align2) + my DRC
config file(s):
https://www.dropbox.com/sh/wvjp38mzahoe77s/AAAFhDU7FvkctmVTapxhXVxCa?dl=0
then the octave graphs based on these IR
https://www.dropbox.com/sh/oi67m08g0wmmkhy/AAAumUlbqkvhG50F2cGA_K6Sa?dl=0
Do you see anything wrong with all this?
Regards
Vincent
2015-04-28 9:19 GMT+02:00 Denis Sbragion <d.sbragion@...>:
> Hello Vincent,
>
> On Mon, April 27, 2015 20:25, Vincent Lejay wrote:
> ...
> > When I say that the measure of the corrected signal is different from the
> > theoretical one, I mainly speak of the "time correction". The frequency
> > response seems good, but thevarious ocatve graphs showing the IR, step
> > reponse... seem more dirty than the original.
>
> it should be the same thing. As long as you don't move the mic the measured
> and simulated IR should be pretty much identical.
>
> Here it is a test I did more than 10 years ago:
>
> http://servizi.infotecna.it/MisVsSim/MisVsSim.png
>
> This has been done using MLS instead of the log sweep as the measurement
> method, and a really cheap SoundBlaster PCI 128 soundcard.
>
> It's not easy to see it but there are actually two tracks there, one blue
> and
> the other one green, almost perfectly superimposed. I don't even remember
> which is the measurements and which is the simulation. Consider that MLS is
> much less accurate than the log sweep, so by using the log sweep and some
> good
> equipment the match should be even better than that.
>
> Bye,
>
> --
> Dr. Denis Sbragion
> InfoTecna
> Tel: +39 0362 805396, Fax: +39 0362 805404
> URL: http://www.infotecna.it
>
>
>
> ------------------------------------------------------------------------------
> One dashboard for servers and applications across Physical-Virtual-Cloud
> Widest out-of-the-box monitoring support with 50+ applications
> Performance metrics, stats and reports that give you Actionable Insights
> Deep dive visibility with transaction tracing using APM Insight.
> http://ad.doubleclick.net/ddm/clk/290420510;117567292;y
> _______________________________________________
> Drc-fir-users mailing list
> Drc-fir-users@...
> https://lists.sourceforge.net/lists/listinfo/drc-fir-users
>

Hello Vincent,
On Mon, April 27, 2015 20:25, Vincent Lejay wrote:
...
> When I say that the measure of the corrected signal is different from the
> theoretical one, I mainly speak of the "time correction". The frequency
> response seems good, but thevarious ocatve graphs showing the IR, step
> reponse... seem more dirty than the original.
it should be the same thing. As long as you don't move the mic the measured
and simulated IR should be pretty much identical.
Here it is a test I did more than 10 years ago:
http://servizi.infotecna.it/MisVsSim/MisVsSim.png
This has been done using MLS instead of the log sweep as the measurement
method, and a really cheap SoundBlaster PCI 128 soundcard.
It's not easy to see it but there are actually two tracks there, one blue and
the other one green, almost perfectly superimposed. I don't even remember
which is the measurements and which is the simulation. Consider that MLS is
much less accurate than the log sweep, so by using the log sweep and some good
equipment the match should be even better than that.
Bye,
--
Dr. Denis Sbragion
InfoTecna
Tel: +39 0362 805396, Fax: +39 0362 805404
URL: http://www.infotecna.it

Hello Vincent,
Are you sure you're not using a minimum phase version of the filter ?You can use the public folder of dropbox or hubic to share files.
Matthieu
Le Lundi 27 avril 2015 20h25, Vincent Lejay <vincent.lejay@...> a écrit :
Hello
I am not at home at the moment (in holidays) but of course I will provide my IR and measured sweeps
Is there a shared place where I can put them?
When I say that the measure of the corrected signal is different from the theoretical one, I mainly speak of the "time correction". The frequency response seems good, but thevarious ocatve graphs showing the IR, step reponse... seem more dirty than the original.
BR
Vincent
2015-04-27 12:36 GMT+02:00 Denis Sbragion <d.sbragion@...>:
Hello Vincent,
sorry, but I missed your e-mail. Some quick answers, more will follow.
On Thu, April 23, 2015 13:34, Vincent Lejay wrote:
...
> I use an ECM8000 mic placed at the listening position, directed toward the
> ceiling (is it right to do like that?). I use as mic preamp a behringer
the ECM8000 isn't perfectly omnidirectional. Better to point it to the middle
of the speaker, but don't expect a big difference.
> I have read through the list that a reference channel can be used to
> correct the timing of the recording chain but I don?t know how to do that.
Least of the problem at this stage, and most of the times not a problem at
all, even for advanced uses.
> Here come most of my questions but I?ll try to be quick. Denis provides
> with his software a set of standard parameters but I am convinced that
> applying optimized parameters for your personal system (incl. the room) can
> provide much better results.
Indeed optimizing the parameters might provide some improvements, but they
aren't as big as you might expect and it is far from easy to do the tuning.
Start by obtaining some good result with some of the standard templates,
starting with soft or normal. After you get it working you might consider
tuning them.
> However like St Thomas I would like to measure the results in order to
> validate the improvements and compare to the theoretical correction given
> by DRC
If you don't move the mic usually a corrected measurement is within a fraction
of a dB from the simulated one. If there are differences then there must be
some mistake somewhere. If this is the case providing both the measured
impulse response and the measured sweeps might be of help in identifying the
cause.
>From your descriptions BTW it looks like DRC isn't working as it should at the
moment.
Bye,
--
Denis Sbragion
InfoTecna
Tel: +39 0362 805396, Fax: +39 0362 805404
URL: http://www.infotecna.it
------------------------------------------------------------------------------
One dashboard for servers and applications across Physical-Virtual-Cloud
Widest out-of-the-box monitoring support with 50+ applications
Performance metrics, stats and reports that give you Actionable Insights
Deep dive visibility with transaction tracing using APM Insight.
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------------------------------------------------------------------------------
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Widest out-of-the-box monitoring support with 50+ applications
Performance metrics, stats and reports that give you Actionable Insights
Deep dive visibility with transaction tracing using APM Insight.
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Hello
I am not at home at the moment (in holidays) but of course I will provide
my IR and measured sweeps
Is there a shared place where I can put them?
When I say that the measure of the corrected signal is different from the
theoretical one, I mainly speak of the "time correction". The frequency
response seems good, but thevarious ocatve graphs showing the IR, step
reponse... seem more dirty than the original.
BR
Vincent
2015-04-27 12:36 GMT+02:00 Denis Sbragion <d.sbragion@...>:
> Hello Vincent,
>
> sorry, but I missed your e-mail. Some quick answers, more will follow.
>
> On Thu, April 23, 2015 13:34, Vincent Lejay wrote:
> ...
> > I use an ECM8000 mic placed at the listening position, directed toward
> the
> > ceiling (is it right to do like that?). I use as mic preamp a behringer
>
> the ECM8000 isn't perfectly omnidirectional. Better to point it to the
> middle
> of the speaker, but don't expect a big difference.
>
> > I have read through the list that a reference channel can be used to
> > correct the timing of the recording chain but I don?t know how to do
> that.
>
> Least of the problem at this stage, and most of the times not a problem at
> all, even for advanced uses.
>
> > Here come most of my questions but I?ll try to be quick. Denis provides
> > with his software a set of standard parameters but I am convinced that
> > applying optimized parameters for your personal system (incl. the room)
> can
> > provide much better results.
>
> Indeed optimizing the parameters might provide some improvements, but they
> aren't as big as you might expect and it is far from easy to do the tuning.
> Start by obtaining some good result with some of the standard templates,
> starting with soft or normal. After you get it working you might consider
> tuning them.
>
> > However like St Thomas I would like to measure the results in order to
> > validate the improvements and compare to the theoretical correction given
> > by DRC
>
> If you don't move the mic usually a corrected measurement is within a
> fraction
> of a dB from the simulated one. If there are differences then there must be
> some mistake somewhere. If this is the case providing both the measured
> impulse response and the measured sweeps might be of help in identifying
> the
> cause.
>
> >From your descriptions BTW it looks like DRC isn't working as it should
> at the
> moment.
>
> Bye,
>
> --
> Denis Sbragion
> InfoTecna
> Tel: +39 0362 805396, Fax: +39 0362 805404
> URL: http://www.infotecna.it
>
>
>
> ------------------------------------------------------------------------------
> One dashboard for servers and applications across Physical-Virtual-Cloud
> Widest out-of-the-box monitoring support with 50+ applications
> Performance metrics, stats and reports that give you Actionable Insights
> Deep dive visibility with transaction tracing using APM Insight.
> http://ad.doubleclick.net/ddm/clk/290420510;117567292;y
> _______________________________________________
> Drc-fir-users mailing list
> Drc-fir-users@...
> https://lists.sourceforge.net/lists/listinfo/drc-fir-users
>

Hello Vincent,
sorry, but I missed your e-mail. Some quick answers, more will follow.
On Thu, April 23, 2015 13:34, Vincent Lejay wrote:
...
> I use an ECM8000 mic placed at the listening position, directed toward the
> ceiling (is it right to do like that?). I use as mic preamp a behringer
the ECM8000 isn't perfectly omnidirectional. Better to point it to the middle
of the speaker, but don't expect a big difference.
> I have read through the list that a reference channel can be used to
> correct the timing of the recording chain but I don?t know how to do that.
Least of the problem at this stage, and most of the times not a problem at
all, even for advanced uses.
> Here come most of my questions but I?ll try to be quick. Denis provides
> with his software a set of standard parameters but I am convinced that
> applying optimized parameters for your personal system (incl. the room) can
> provide much better results.
Indeed optimizing the parameters might provide some improvements, but they
aren't as big as you might expect and it is far from easy to do the tuning.
Start by obtaining some good result with some of the standard templates,
starting with soft or normal. After you get it working you might consider
tuning them.
> However like St Thomas I would like to measure the results in order to
> validate the improvements and compare to the theoretical correction given
> by DRC
If you don't move the mic usually a corrected measurement is within a fraction
of a dB from the simulated one. If there are differences then there must be
some mistake somewhere. If this is the case providing both the measured
impulse response and the measured sweeps might be of help in identifying the
cause.
>From your descriptions BTW it looks like DRC isn't working as it should at the
moment.
Bye,
--
Denis Sbragion
InfoTecna
Tel: +39 0362 805396, Fax: +39 0362 805404
URL: http://www.infotecna.it

Hi Vincent,
Normally the measurements after correction are very close to the predicted ones.The best would be to share plots and raw measurements :)
Matthieu
Le Jeudi 23 avril 2015 13h34, Vincent Lejay <vincent.lejay@...> a écrit :
Dear DRC-FIR users
First a small introduction: I am a frenchaudio/hifi enthusiast, I am an engineer in computer science though I have nospecific knowledge in signal processing.Thank you all and especially Denis forproviding all his knowledge in such a great tool, and all for free.I have read with interest a good part ofthe messages of this list. I have to admit that I lack basics in acoustics andsignal processing to correctly understand everything but I feel interested tolearn more and to apply the DRC principles to my own system.I have been “playing” with DRC FIR for acouple of months now. A lot of questions came to me, probably because I am notan expert. What I would like here is to ask some them and if possible validatemy measurements and my DRC config file with those who wish to help me.My system is composed of a PC (window XP) withfoobar (+convolver plugin). My sound card is a lynx 2B. I use it for recordingand Home theatre playback. For stereo playback I use an external DAC which isslightly better than the lynx. My speakers are verity audio fidelio. My room isa rectangle of 7m * 4m. Due to the room configuration (doors, windows, furniture…)the speakers are placed along the longer wall, quite centered on it, a littleon the right side. The listening position is on the opposite wall.My purpose using DRC is to correct as muchas possible the artefacts in the sound reproduction introduced by the room, andnot to correct the speakers, assuming that they have been properly designed.
1. RecordingI know that recording is a key element ofthe process because correcting a measure that does not reflect the real systemwould probably be worse than doing nothing.I use an ECM8000 mic placed at thelistening position, directed toward the ceiling (is it right to do like that?).I use as mic preamp a behringer mic800 with no effect, which is connected tothe lynx2. I use align2 software (from Jean Luc Ohl) to record the logsweep playbackand provide the system impulse response.I may provide my measurement files if someof you are interested to help me validating that they are suitable as input forDRC processingI have read through the list that areference channel can be used to correct the timing of the recording chain butI don’t know how to do that.
2. DRC parameters.Here come most of my questions but I’ll tryto be quick. Denis provides with his software a set of standard parameters butI am convinced that applying optimized parameters for your personal system (incl.the room) can provide much better results.I would classify the .drc parameters in twosets:- Those related to the FDW- Those related parameters of thealgorithms applied at the different steps (such as choice of the algorithm, minimumphase, linear phase, disabling some steps…) Concerning the FDW:After reading the messages posted in thislist I feel quite puzzled because DRC-FIR is based on very precise acousticsand mathematics laws but on the other hand the suggested way to configure it isvery empirical and subjective, I mean it seems there is no other method thangenerating various filters with sets of different parameters, then listeningand detecting potential artefacts. Since I don’t trust very much my ability todetermine the good correction with my only ears, I have tried to use octavegraphs to determine the size of the correction window which is applicable to mysystem. To do that I have slightly modified the spectrogram 20ms graph providedwith DRC package in order to:Change the colour range from 0 to -12db(instead of -90db),Display on the graph the frequencydependent curve My idea is to trace the FDW on the spectrogramgraph with different UpperWindow, LowerWindow, WindowExponent parameters and inthe end choose the parameters which leave the curve the closest at the right ofthe red and yellow colours on the spectrogram (which I assume represent thebigger part of the direct sound – with less than 12db attenuation).In my system it gives a lowerwindow closeto 200ms, an upperwindow close to 10ms and an exponent close to 1.0; theupperwindow size is a big question for me as I expected a much lower figure.This modified spectrogram graph is myintuitive approach to set parameters in DRC config file based on real measureddata. However I am not sure at all that my strategy is correct, especially Iwonder in which of the DRC filters computation steps my resulting parameters wouldbe applicable. Today I used them for the FDW in the minphase stage. But becauseof my poor understanding of what is the minphase and the excess phase I fear todo things the wrong way. So if you have a better method it take it!
Concerning the algorithmsAs a first approach I would tend to stickto those present in the default DRC config files. However I am curious about the effect ofchanging some of them, especially concerning the minimum phase or linear phasealgorithms. I certainly missed something but I wonder why some steps of thefull DRC correction apply minimum phase algorithm: if one of the main goal ofDRC is to correct phase problems, then min phase algorithms go on the oppositedirection, don’t they?In my .drc config file, I have disabled themic compensation step (after reading messages from this list, it seemshazardous to use a predefined file), and the psychoacoustic step. I have builta target response curve which follows as much as possible the natural responseof my speakers, with a limitation of +- 3db away from a flat response. 3. ResultsResults are for sure audible, the precisionof the scene seems better, the centering seems better.However like St Thomas I would like tomeasure the results in order to validate the improvements and compare to thetheoretical correction given by DRCTo do that I have used the “convert” optionof foobar which allows to convert audio files from a format to another and inthe same time apply DSP. I have converted the sweep file .wav to .wav withapplication of the convolution filter generated with DRC. Then I have playedthe sweep file with align2 (same procedure as for the initial measurements) anddrawn with octave the graphs of the obtained IR. At the moment I am notsatisfied with the obtained graphs. What they show is very different from thetheoretical IR response (rtc.pcm generated by DRC on the original IR). Thismakes me think that there is something wrong somewhere, either in the roomitself, either in my measure or correction procedures. This is why I decided toask some advice on this list, in order to detect where the things go wrong andwhat to do to improve my method.I can provide any of my files if you wish.Thank you for reading my message and itwill be a pleasure to be in contact with some of you.
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Dear DRC-FIR users
First a small introduction: I am a french audio/hifi enthusiast, I am an
engineer in computer science though I have no specific knowledge in signal
processing.
Thank you all and especially Denis for providing all his knowledge in such
a great tool, and all for free.
I have read with interest a good part of the messages of this list. I have
to admit that I lack basics in acoustics and signal processing to correctly
understand everything but I feel interested to learn more and to apply the
DRC principles to my own system.
I have been “playing” with DRC FIR for a couple of months now. A lot of
questions came to me, probably because I am not an expert. What I would
like here is to ask some them and if possible validate my measurements and
my DRC config file with those who wish to help me.
My system is composed of a PC (window XP) with foobar (+convolver plugin).
My sound card is a lynx 2B. I use it for recording and Home theatre
playback. For stereo playback I use an external DAC which is slightly
better than the lynx. My speakers are verity audio fidelio. My room is a
rectangle of 7m * 4m. Due to the room configuration (doors, windows,
furniture…) the speakers are placed along the longer wall, quite centered
on it, a little on the right side. The listening position is on the
opposite wall.
My purpose using DRC is to correct as much as possible the artefacts in the
sound reproduction introduced by the room, and not to correct the speakers,
assuming that they have been properly designed.
1. Recording
I know that recording is a key element of the process because correcting a
measure that does not reflect the real system would probably be worse than
doing nothing.
I use an ECM8000 mic placed at the listening position, directed toward the
ceiling (is it right to do like that?). I use as mic preamp a behringer
mic800 with no effect, which is connected to the lynx2. I use align2
software (from Jean Luc Ohl) to record the logsweep playback and provide
the system impulse response.
I may provide my measurement files if some of you are interested to help me
validating that they are suitable as input for DRC processing
I have read through the list that a reference channel can be used to
correct the timing of the recording chain but I don’t know how to do that.
2. DRC parameters.
Here come most of my questions but I’ll try to be quick. Denis provides
with his software a set of standard parameters but I am convinced that
applying optimized parameters for your personal system (incl. the room) can
provide much better results.
I would classify the .drc parameters in two sets:
- Those related to the FDW
- Those related parameters of the algorithms applied at the different steps
(such as choice of the algorithm, minimum phase, linear phase, disabling
some steps…)
*Concerning the FDW:*
After reading the messages posted in this list I feel quite puzzled because
DRC-FIR is based on very precise acoustics and mathematics laws but on the
other hand the suggested way to configure it is very empirical and
subjective, I mean it seems there is no other method than generating
various filters with sets of different parameters, then listening and
detecting potential artefacts. Since I don’t trust very much my ability to
determine the good correction with my only ears, I have tried to use octave
graphs to determine the size of the correction window which is applicable
to my system. To do that I have slightly modified the spectrogram 20ms
graph provided with DRC package in order to:
Change the colour range from 0 to -12db (instead of -90db),
Display on the graph the frequency dependent curve
My idea is to trace the FDW on the spectrogram graph with different
UpperWindow, LowerWindow, WindowExponent parameters and in the end choose
the parameters which leave the curve the closest at the right of the red
and yellow colours on the spectrogram (which I assume represent the bigger
part of the direct sound – with less than 12db attenuation).
In my system it gives a lowerwindow close to 200ms, an upperwindow close to
10ms and an exponent close to 1.0; the upperwindow size is a big question
for me as I expected a much lower figure.
This modified spectrogram graph is my intuitive approach to set parameters
in DRC config file based on real measured data. However I am not sure at
all that my strategy is correct, especially I wonder in which of the DRC
filters computation steps my resulting parameters would be applicable.
Today I used them for the FDW in the minphase stage. But because of my poor
understanding of what is the minphase and the excess phase I fear to do
things the wrong way. So if you have a better method it take it!
*Concerning the algorithms*
As a first approach I would tend to stick to those present in the default
DRC config files.
However I am curious about the effect of changing some of them, especially
concerning the minimum phase or linear phase algorithms. I certainly missed
something but I wonder why some steps of the full DRC correction apply
minimum phase algorithm: if one of the main goal of DRC is to correct phase
problems, then min phase algorithms go on the opposite direction, don’t
they?
In my .drc config file, I have disabled the mic compensation step (after
reading messages from this list, it seems hazardous to use a predefined
file), and the psychoacoustic step. I have built a target response curve
which follows as much as possible the natural response of my speakers, with
a limitation of +- 3db away from a flat response.
3. Results
Results are for sure audible, the precision of the scene seems better, the
centering seems better.
However like St Thomas I would like to measure the results in order to
validate the improvements and compare to the theoretical correction given
by DRC
To do that I have used the “convert” option of foobar which allows to
convert audio files from a format to another and in the same time apply
DSP. I have converted the sweep file .wav to .wav with application of the
convolution filter generated with DRC. Then I have played the sweep file
with align2 (same procedure as for the initial measurements) and drawn with
octave the graphs of the obtained IR. At the moment I am not satisfied with
the obtained graphs. What they show is very different from the theoretical
IR response (rtc.pcm generated by DRC on the original IR). This makes me
think that there is something wrong somewhere, either in the room itself,
either in my measure or correction procedures. This is why I decided to ask
some advice on this list, in order to detect where the things go wrong and
what to do to improve my method.
I can provide any of my files if you wish.
Thank you for reading my message and it will be a pleasure to be in contact
with some of you.

Hi Gregory
Does that include distortion analysis, or just the impulse response? Can you point me to the relevant code?
Matthias
----------
http://audioroot.net – custom audio equipment
> On 12 Jan 2015, at 21:39, Gregory Maxwell <gmaxwell@...> wrote:
>
> Yes, DRC-fir includes a tool for log-swept-sine measurements.
>
> On Mon, Jan 12, 2015 at 8:36 PM, audioroot <matthias@...> wrote:
>> Hi all
>>
>> I am new to this list, so please be patient with me…
>>
>> I am trying the implement the "Farina sine sweep method” for analysis of harmonic distortion in my own software using (MATAA, see http://audioroot.net/mataa-mats-audio-analyzer ). I read the Farina papers, but I was unable to reproduce the method from this. I am therefore looking for a worked example or source code that implements the Farina method. Is there something like this in DRC-FIR, or can someone point me in the right direction?
>>
>> Matthias
>>
>> ----------
>> http://audioroot.net – custom audio equipment
>>
>>
>>
>>
>>
>> ------------------------------------------------------------------------------
>> New Year. New Location. New Benefits. New Data Center in Ashburn, VA.
>> GigeNET is offering a free month of service with a new server in Ashburn.
>> Choose from 2 high performing configs, both with 100TB of bandwidth.
>> Higher redundancy.Lower latency.Increased capacity.Completely compliant.
>> http://www.gigenet.com
>> _______________________________________________
>> Drc-fir-users mailing list
>> Drc-fir-users@...
>> https://lists.sourceforge.net/lists/listinfo/drc-fir-users
>
> ------------------------------------------------------------------------------
> New Year. New Location. New Benefits. New Data Center in Ashburn, VA.
> GigeNET is offering a free month of service with a new server in Ashburn.
> Choose from 2 high performing configs, both with 100TB of bandwidth.
> Higher redundancy.Lower latency.Increased capacity.Completely compliant.
> http://www.gigenet.com
> _______________________________________________
> Drc-fir-users mailing list
> Drc-fir-users@...
> https://lists.sourceforge.net/lists/listinfo/drc-fir-users

Yes, DRC-fir includes a tool for log-swept-sine measurements.
On Mon, Jan 12, 2015 at 8:36 PM, audioroot <matthias@...> wrote:
> Hi all
>
> I am new to this list, so please be patient with me…
>
> I am trying the implement the "Farina sine sweep method” for analysis of harmonic distortion in my own software using (MATAA, see http://audioroot.net/mataa-mats-audio-analyzer ). I read the Farina papers, but I was unable to reproduce the method from this. I am therefore looking for a worked example or source code that implements the Farina method. Is there something like this in DRC-FIR, or can someone point me in the right direction?
>
> Matthias
>
> ----------
> http://audioroot.net – custom audio equipment
>
>
>
>
>
> ------------------------------------------------------------------------------
> New Year. New Location. New Benefits. New Data Center in Ashburn, VA.
> GigeNET is offering a free month of service with a new server in Ashburn.
> Choose from 2 high performing configs, both with 100TB of bandwidth.
> Higher redundancy.Lower latency.Increased capacity.Completely compliant.
> http://www.gigenet.com
> _______________________________________________
> Drc-fir-users mailing list
> Drc-fir-users@...
> https://lists.sourceforge.net/lists/listinfo/drc-fir-users

Hi all
I am new to this list, so please be patient with me…
I am trying the implement the "Farina sine sweep method” for analysis of harmonic distortion in my own software using (MATAA, see http://audioroot.net/mataa-mats-audio-analyzer ). I read the Farina papers, but I was unable to reproduce the method from this. I am therefore looking for a worked example or source code that implements the Farina method. Is there something like this in DRC-FIR, or can someone point me in the right direction?
Matthias
----------
http://audioroot.net – custom audio equipment

Hello Daniel,
On Tue, May 20, 2014 11:04, Daniel Arteaga wrote:
> At short filter sizes the filter is showing a low-frequency roll-off, so
> that it doesn't follow the target at low frequencies. I have tried
> varying several parameters and the result is always the same.
how strong is it? Do you have some graph? BTW, depending on the target and the
length, this might be a correct behaviour.
> I understand that with short filters it is difficult to control the
> behvaviour at low frequencies, but, is there a way to at least
> compensate or workaround this effect?
Windowing is the theoretical right approach. You might try different windows,
for example by importing the filter into Octave/Matlab and using the windows
available there. But don't expect much of a difference. There's the usual
Gabor inequality setting the rules here, so if it is short it can't have high
resolution in the frequency domain.
Bye,
--
Denis Sbragion
InfoTecna
Tel: +39 0362 805396, Fax: +39 0362 805404
URL: http://www.infotecna.it

Hi again Denis, hello all,
El 27/03/14 12:57, Denis Sbragion ha escrit:
> Hello Daniel,
>
> On Thu, March 27, 2014 11:09, Daniel Arteaga wrote:
> ...
>> How can we reduce the size of final filters only, without reducing the
>> overall quality of the filter?
>
> the best option is to change only the final window size, i.e. PSOutWindow for
> the mixed phase filter and MSOutWindow for the minimum phase filter. Set also
> MSFilterDelay to 0 with the minimum phase filter so that all the available
> filter length is used for the correction.
At short filter sizes the filter is showing a low-frequency roll-off, so
that it doesn't follow the target at low frequencies. I have tried
varying several parameters and the result is always the same.
I'm guessing that this might be an effect of the final windowing that
introduces some kind of high-pass filtering at low frequencies.
I understand that with short filters it is difficult to control the
behvaviour at low frequencies, but, is there a way to at least
compensate or workaround this effect?
Thank you very much,
Daniel

Hi Denis, hi all,
Is it possible to tell DRC not to correct a given frequency region?
Assume we want to correct only over 200 Hz, and leave from 20 to 200 Hz
the filter flat.
If I replace all ??StartFreq values from 20 to 200 I get a filter which
is not flat until 200 Hz but rather has some LF rolloff. Is it possible
to get the flat filter behaviour for the uncorrected region?
Thank you very much,
Daniel

El 27/03/14 12:57, Denis Sbragion ha escrit:
> Hello Daniel,
>
> On Thu, March 27, 2014 11:09, Daniel Arteaga wrote:
> ...
>> How can we reduce the size of final filters only, without reducing the
>> overall quality of the filter?
>
> the best option is to change only the final window size, i.e. PSOutWindow for
> the mixed phase filter and MSOutWindow for the minimum phase filter. Set also
> MSFilterDelay to 0 with the minimum phase filter so that all the available
> filter length is used for the correction.
>
> If you plan to use a mixed phase correction it would be also a good idea to
> reduce the excess phase correction in order to leave most of the taps to the
> minimum phase part, but this would require a major rewriting of the whole
> template. Or just drop the mixed phase correction altogether and stick with
> the minimum phase one. With this filter sizes it might be a better option.
Thank you very much Denis!
Daniel

Hello Daniel,
On Thu, March 27, 2014 11:09, Daniel Arteaga wrote:
...
> How can we reduce the size of final filters only, without reducing the
> overall quality of the filter?
the best option is to change only the final window size, i.e. PSOutWindow for
the mixed phase filter and MSOutWindow for the minimum phase filter. Set also
MSFilterDelay to 0 with the minimum phase filter so that all the available
filter length is used for the correction.
If you plan to use a mixed phase correction it would be also a good idea to
reduce the excess phase correction in order to leave most of the taps to the
minimum phase part, but this would require a major rewriting of the whole
template. Or just drop the mixed phase correction altogether and stick with
the minimum phase one. With this filter sizes it might be a better option.
Bye,
--
Denis Sbragion
InfoTecna
Tel: +39 0362 805396, Fax: +39 0362 805404
URL: http://www.infotecna.it

Hi all,
We would need to design a short correction filter, of 1024 or 2048 taps.
We are using a modified minimal drc configuration files, so that even
with the standard filter length of 65536 the "real" length of the filter
is already, quite short, of the order of 1024 taps.
We tried replaning all instances of the several ??FilterLen parameters
by 1024, but the quality of the final filter is much worse. I understand
that this is an effect of truncating intermediate filters which are
possibly much longer.
How can we reduce the size of final filters only, without reducing the
overall quality of the filter?
Thank you very much,
--
Daniel Arteaga
Scientific Director - Audio
Barcelona Media
Av. Diagonal 177, 08018 Barcelona
Tel: (+34) 932 381 400
http://www.barcelonamedia.org

Hello Ignacio,
On Sat, January 4, 2014 00:25, Ignacio Sabido wrote:
> This link is broken:
>
> - rec_imp:
> http://www.duffroomcorrection.com/wiki/Simple_Automated_IR_Measuring_Tool
>
>>From where I could download the latest version of Rec_imp?
yep, the duffroomcorrection site seems to be down since quite a long time. The
rec_imp tool was maintained by Ed Wildgoose, which was also the owner of the
web site. You should try to contact him for any information. The
http://www.archive.org wayback machine may be another option.
Bye,
--
Dr. Denis Sbragion
InfoTecna
Tel: +39 0362 805396, Fax: +39 0362 805404
URL: http://www.infotecna.it

HI again Denis!!
This link is broken:
- rec_imp:
http://www.duffroomcorrection.com/wiki/Simple_Automated_IR_Measuring_Tool
>From where I could download the latest version of Rec_imp?
Thanks in advaced!
2012/8/12 Denis Sbragion <d.sbragion@...>
> On Sat, August 11, 2012 16:48, Nicola Berndt wrote:
> > might actually be, although REW normally complains when it has the
> > feeling something is wrong with the timing. If I look at the Impulse in
> > REW it looks somewhat bad, but I would not know what to make of it. See
> > here: http://komeda-berlin.de/images/drc/08/impulse_widerange_r.jpg
>
> :O Are you sure that measurement is correct? I have little to no
> experience of
> car environment, but it's hard to believe that it could generate somthing
> like
> that.
>
> Have you tried some loopback measurement just to check that everything is
> correct?
>
> Bye,
>
> --
> Denis Sbragion
> InfoTecna
> Tel: +39 0362 805396, Fax: +39 0362 805404
> URL: http://www.infotecna.it
>
>
>
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>

Hello Jean-Luc,
On Sun, June 16, 2013 09:19, Jean-Luc Ohl wrote:
> different normalisation types are proposed for the calculated filter
> (chapter 4.6.2 in manual) but how do you calculate the max gain of this
> filter ?
> Can you do it directly from the FIR coefficients ? Or from a FFT and
> then check magnitude ? Or ?
> I'm curious about how you did it.
quite in a hurry, but if memory serves me, it was just max of the filter
magnitude reponse. Unless the target is far from flat this is essentially the
same as reducing the filter level by the maximum allowed filter gain (6 dB by
default). So it is pretty much automation of the obvious. :)
Bye,
--
Denis Sbragion
InfoTecna
Tel: +39 0362 805396, Fax: +39 0362 805404
URL: http://www.infotecna.it

Hi all,
a question especially for Denis :
different normalisation types are proposed for the calculated filter
(chapter 4.6.2 in manual) but how do you calculate the max gain of this
filter ?
Can you do it directly from the FIR coefficients ? Or from a FFT and
then check magnitude ? Or ?
I'm curious about how you did it.
Have a nice day
Jean-Luc

Hello Alan,
On Sat, June 15, 2013 03:40, Alan Jordan wrote:
...
> Has anyone been able to compile rec_imp on a 64 bit version of Ubuntu? If
> not, what are people using these days for recording the impulse response
> under Linux?
under linux I always used aplay/arec along with glsweep, lsconv and some shell
script glue. Don't know if, for example, SoX is able to do the work of the
arec/aplay pair under Windows. Once it wasn't, but it was quite some times
ago.
Bye,
--
Denis Sbragion
InfoTecna
Tel: +39 0362 805396, Fax: +39 0362 805404
URL: http://www.infotecna.it

Hello,
I am trying to configure DRCDesigner to be able to run under the newer 64
bit Linux releases, but cannot get rec_imp to compile on a 64 bit system.
Has anyone been able to compile rec_imp on a 64 bit version of Ubuntu? If
not, what are people using these days for recording the impulse response
under Linux?
Thanks,
Alan